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300-075 exam Dumps Source : Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)
Test Code : 300-075
Test appellation : Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)
Vendor appellation : Cisco
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Cisco's IP telephony offering has become so complicated in such a brief length of time that to find out about it now includes a two-part, eight-day working towards route. Delegates can both attend the two parts as a standalone option (and pick the course exam if they opt for), or pick a further four lessons and fade for the Cisco certified Voice professional (CCVP) qualification.
global competencies is an reliable practicing companion of Cisco, and so intently follows its curriculum. It is among the largest training suppliers for Cisco in the UK, having acquired the ragged biggest company, Azlan, in March 2006. because of the acquisition, it boasts a purview of working towards centres throughout the South East, the Midlands and the North.
The course specializes in CallManager (now called Cisco Unified Communications manager), the call-processing software in Cisco's IP telephony providing. Delegates can choose to study CallManager edition 4, the most widely deployed windows-based edition, or edition 5, which is Linux-based mostly.
ZDNet.co.uk reviewed the first a fraction of the eight-day course and chose to focal point on edition four. the first fraction lasts 5 days and covers about five-eighths of the content. Delegates are given dates for the 2d half, which is three days in size, presently after the conclusion of fraction one.
LocationWe reviewed this course at world potential's London training centre, a diligent basement facility in the heart of the metropolis's enterprise district. international odds besides teaches the route in its Wakefield, Coventry and Wokingham working towards centres.
PrerequisitesGlobal odds suggests that delegates maintain already studied for each the Cisco certified Networking associate (CCNA) and Cisco Voice over IP (CVoice) certifications. however, basically, a number of years of journey in both statistics and voice communications is probably going to subsist sufficient.
topics coveredCallManager working gadget, database and supporting functions; CallManager deployment alternatives; setting up a call; Cisco telephony endpoints; CallManager configuration; configuring leaven switches to wield voice; configuring Cisco IP Communicator gentle phone; the usage of the bulk administration device; configuring gateways and trunks; configuring route plans; configuring hunt groups; imposing category of handle; multi-website deployments; enabling points; configuring CallManager purposes; adding video; first-rate of provider; protection; troubleshooting.
DeliveryThis path changed into delivered by using an extremely knowledgeable tutor who understood Cisco's IP telephony gear fully. He changed into hardly ever unable to reply a question and he populated the course with factual titbits, some of that maintain been positive in functional phrases ('bear in sarcasm that passwords are reset when a CallManager update is applied') and some of that maintain been just simple wonderful ('did you know why the MAC wield carries that prefix?'). significantly, he gave close pointers to the category of questions that may loom within the examination.
despite the fact the tutor is permanently employed by world talents, he said that he spends 20 percent of his working time outside the school margin so as to sustain-to-date with IP and Cisco expertise. His title appeared to subsist substantiated, as he talked at size about trade hobbies and functional experiences.
The tutor besides ran useful workouts configuring CallManager, which accounted for approximately 60 p.c of the route's period. each and every step was clearly described in precise detail in the course notes, with the influence that dinky thought was required to follow the notes, arguably impeding the discovering manner.
We had only 1 criticism of the tutor: however he held the delegates' attention admirably, he did, during this reviewer's opinion, fail to wholly engage with the delegates in two-means communication.
best of aiding materialsDelegates were provided with three hefty units of notes overlaying the tutor's shows and useful workouts. These had been for essentially the most fraction provided through Cisco, with little local additions. this is typical observe for a technology training course. Delegates maintain been besides offered free ongoing lead with the aid of electronic mail.
cost-addOur judgement on the charge of the path for odds delegates comes with one caveat. Delegates ought to accept as just with the chance that they're going to deserve to install an IP telephony providing it really is made by Cisco. in terms of configuration, Cisco's IP telephony machine is significantly distinctive to any other seller's device, and so the path handiest has tremendous charge for Cisco deployments. IT specialists who're planning to install non-Cisco IP telephony could conclude more desirable in deciding on one of the many widely wide-spread IP-telephony classes on present from different suppliers.
however, if the delegate is import to deploy Cisco package, then the price-add from the eight days is considerable: an vital amount of hands-on potential may besides subsist taken away.
Certification and additional opportunitiesDelegates can determine to sit down an examination after the discontinue of the direction if they hope, and might then proceed to complete the CCVP if they desire. The other lessons inside the CCVP are every sole provided via international skills. they are: worthy of provider; Cisco Voice over IP; Troubleshooting Cisco Unified Communications methods; enforcing Cisco Voice Gateways and Gatekeepers.
however a notice of warning: the total fees to attend every sole of the classes faultfinding for the CCVP will set delegates again a color over £10,000 — and that is the reason devoid of examination prices or VAT. The certification can besides subsist held for 3 years before re-testing is required.
students might then choose to analyze for the much sought-after certified Cisco Internetwork skilled (CCIE) qualification, Cisco's best certification.
Citing stories claiming billion greenback boom in on-demand media, Cisco has announced a new certification for affluent media in IP networks.
in keeping with the company, its new certification, affluent Media Communications specialist, will subsist confident IT professionals "have the practising to forward the skills required to design, deploy and manage prosperous media functions to wield new market calls for, including integrating video conferencing functions right into a voice and records network."
Cisco items the certification encompasses will consist of Cisco CallManager for IP telephony, Cisco harmony voice messaging and Cisco IP Contact facilities.
earning a affluent Media professional certification will require conserving an lively Cisco certified network affiliate (CCNA) certification and passing two extra assessments:
The trade is providing the implementing Cisco IP Video conference wealthy Media Communications (ICRMC) path, a self-paced category it expects most students will finished in about eight hours. The company's informed practicing for CVOICE is attainable as a 5-day instructor-led type, and as a few e-discovering options.
Sao Paulo, Brazil, December eight, 2015 - SLC Agricola, a Brazilian agricultural commodities producer, has adopted Cisco® collaboration technology to upgrade its conversation infrastructure. The enterprise deployed Cisco IP phones, video conferencing solutions and speedy messaging applications to connect its headquarters and 14 construction sites. benefits from the expertise adoption encompass decreased mobilephone and transport expenses besides stronger communications and superior staff productivity.
headquartered in 1977 by pass of the SLC group, SLC Agricola changed into one of the vital first Brazilian organizations specializing in the grains and cotton sector to subsist listed on the stock market, fitting a benchmark in the phase. The enterprise operates 14 farms across six Brazilian states and is in consistent growth, so the ancient telephony platform became saturated. furthermore, lots of the production facilities can subsist found in faraway locations, so the issue in deploying terminals and keeping the analog device became expensive and counterproductive.
The Cisco’s associate Infra TI led the telecomm integration task, and after an entire evaluation on the entire SLC Agricola environment, it designed the migration scheme to the IP system. step one become conducted at SLC's headquarters in Porto Alegre, by pass of implementing Cisco CallManager servers and more than 200 consumer extensions. Two carrier portfolios were built: a typical one with Cisco Unified SIP mobile 3905 contraptions for the team of workers; and a further one for managers and heads of departments, with Cisco Unified IP telephone 8945 capabilities, including brief-start videoconferencing.
The Cisco Jabber® software, which allows mixed rapidly messaging (IM), voice and video, and Cisco freeway were besides carried out to aid using cell gadgets without a VPN.
"the brand new platform has vastly contributed to decreased prices and streamlined production, and we've several examples to prove it," referred to João Aranda, telecom coordinator at SLC Agricola. "To identify one, when the SLC’s presidential group of workers became visiting farm, they were capable of attend an pressing meeting via Cisco video conferencing, saving time and fade back and forth charges, and warding off any distress to the resolution-making manner," he talked about.
"Collaboration options maintain a different utility to agribusiness," talked about Ana Claudia Plihal, industrial director at Cisco. "IP expertise has the vigour to eliminate the exorbitant cabling of established telephony, simplifying the operation in far off environments. At SLC, for instance, the brand new gear will no longer interfere within the farm’s infrastructure. It’s a simple case of plugging new telephones to the gateway of every area," she pointed out. "additionally, the probability to maintain video conferencing available anytime not handiest adds value to the trade but besides presents the enterprise an imaginative profile."
The SLC Agrícola assignment began in 2014, and these days every sole farms already maintain at the least one video conferencing unit apportion in. additional benefits highlighted via the assignment group encompass simplified administration and adjustments in the corporate lifestyle, which directly influenced the resolution-making technique.
Aranda delivered that discounts made on the headquarters, production units and by the organization’s executives with decreased mobilephone and shuttle prices had been no longer calculated yet; although, the consequences completed with the brand new IP expertise are gigantic. "we maintain actually had considerable positive aspects…. for example, many of their internal procedures are now obtainable by pass of IP telephony, comparable to workforce practising and the organization’s trainee program.” SLC Agricola’s telecommunications help is anticipated to subsist accomplished inside two to three years.
About SLC Agricola
SLC Agricola, is an agricultural producer, focusing chiefly on cotton, soybeans and corn. The trade operates 14 farms strategically determined in six Brazilian states totaling 343.6 thousand hectares within the 2013/14 advertising 12 months, and a forecast of planted area of 370.1 thousand hectares for the 2014/15 crop, being 206.9 thousand hectares planted with soybeans, 98.6 thousand with cotton, 43.1 thousand with corn, and 18.1 thousand hectares with other crops reminiscent of espresso, wheat, corn seed and sugarcane. The enterprise’s enterprise model is in line with a big-scale, coincident creation system, with standardized production devices, leading know-how, strict cost controls, and environmental accountability.
Cisco (NASDAQ: CSCO) is the global chief in IT that helps businesses enmesh the alternatives of tomorrow by proving that wonderful issues can pick plot if you connect the up to now unconnected. For ongoing news, tickle fade to http://thenetwork.cisco.com.
# # #
Cisco and the Cisco emblem are emblems or registered emblems of Cisco and/or its affiliates within the U.S. and different countries. a list of Cisco's emblems can subsist discovered at www.cisco.com/go/logos. Third-celebration emblems mentioned are the property of their respective owners. the exercise of the note companion doesn't imply a partnership relationship between Cisco and another enterprise.
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Chapter 1, Identifying Issues in a Multisite Deployment
Deploying Cisco Unified Communications Manager between multiple sites requires an appropriate dial plan, appropriate bandwidth, a character of service (QoS) implementation and a design that can survive IP WAN failures.
This chapter will define the issues that can arise in a multisite Cisco UC Manager deployment, and recommend specific solutions.
Identifying Issues in a Multisite DeploymentTable of contents:1. Multisite deployment challenge overview2. character challenges3. Bandwidth challenges4. Availability challenges5. Dial scheme challenges6. Overlapping and nonconsecutive numbers7. Fixed versus variable-length numbering plans8. Variable-length numbering, E.164 addressing and DID9. Optimized call routing and PSTN backup10. NAT and security issues
Multisite deployment challenge overview
Formerly known as Cisco call Manager (CCM), Cisco Unified Communications Manager (CUCM) multi-site deployment challenges can include the following:
IP networks were not originally designed to carry real-time traffic; instead, they were designed for resiliency and weakness tolerance. Each packet is processed separately in an IP network, sometimes causing different packets in a communications stream to pick different paths to the destination. The different paths in the network may maintain a different amount of packet loss, delay, and delay variation (jitter) because of bandwidth, distance, and congestion differences. The destination must subsist able to receive packets out of order and resequence these packets. This challenge is solved by the exercise of Real-Time Transport Protocol (RTP) sequence numbers and traffic resequencing. When possible, it is best to not depend solely on these RTP mechanisms. Proper network design, using Cisco router Cisco Express Forwarding (CEF) switch cache technology, performs per-destination load sharing by default. Per-destination load sharing is not a perfect load-balancing paradigm, but it ensures that each IP stream (voice call) takes the selfsame path.
Bandwidth is shared by multiple users and applications, whereas the amount of bandwidth required for an individual IP stream varies significantly during short lapses of time. Most data applications are very bursty, whereas Cisco real-time audio communications with RTP exercise the selfsame continuous-bandwidth stream. The bandwidth available for any application, including CUCM and voice-bearer traffic, is unpredictable. During peak periods, packets exigency to subsist buffered in queues waiting to subsist processed because of network congestion. Queuing is a term that anyone who has ever experienced air flight is close with. When you arrive at the airport, you must acquire in a line (queue), because the number of ticket agents (bandwidth) available to check you in is less than the stream of traffic arriving at the ticket counters (incoming IP traffic). If congestion occurs for too long, the queue (packet buffers) gets filled up, and passengers are annoyed (packets are dropped). Higher queuing delays and packet drops are more likely on highly loaded, slow-speed links such as WAN links used between sites in a multisite environment. character challenges are common on these types of links, and you exigency to wield them by implementing QoS. Without the exercise of QoS, voice packets suffer delay, jitter, and packet loss, impacting voice quality. It is faultfinding to properly configure Cisco QoS mechanisms discontinue to discontinue throughout the network for proper audio and video performance.This chapter was excerpted with permissions from Implementing Cisco Unified Communications Manager, fraction 2, by Chris Olsen, Copyright 2008. For more information about this engage and other titles, tickle visit Cisco Press.
During peak periods, packets cannot subsist sent immediately because of interface congestion. Instead, the packets are temporarily stored in a queue, waiting to subsist processed. The amount of time the packet waits in the queue, called the queuing delay, can vary greatly based on network conditions and traffic arrival rates. If the queue is full, newly received packets cannot subsist buffered anymore and acquire dropped (tail drop). pattern 1-1 illustrates tail drop. Packets are processed on a first in, first out (FIFO) model in the hardware queue of every sole router interfaces. Voice conversations are predictable and constant (sampling is every 20 milliseconds by default), but data applications are bursty and greedy. Voice therefore is topic to degradation of character because of delay, jitter, and packet loss.
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Each site in a multisite deployment usually is interconnected by an IP WAN, or occasionally by a metropolitan-area network (MAN) such as Metro Ethernet. Bandwidth on WAN links is limited and relatively expensive. The goal is to exercise the available bandwidth as efficiently as possible. Unnecessary traffic should subsist removed from the IP WAN links through content filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for bandwidth optimization should subsist considered as well. Any period of congestion could result in service degradation unless QoS is deployed throughout the network.
Voice streams are constant and predictable for Cisco audio packets. Typically, the G.729 codec is used across the WAN to best exercise bandwidth. As a comparison, the G.711 audio codec requires 64 kbps, whereas packetizing the G.711 voice sample in an IP/UDP/RTP header every 20 ms requires 16 kbps plus the Layer 2 header overhead.
Voice is sampled every 20 ms, resulting in 50 packets per second (pps). The IP header is 20 bytes, whereas the UDP header is 8 bytes, and the RTP header is 12 bytes. The 40 bytes of header information must subsist converted to bits to pattern out the packet rate of the overhead. Because a byte has 8 bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are sent 50 times per second based on the 20-ms rate (1 millisecond is 1/1000 of a second, and 20/1000 = .02). So:
.02 * 50 = 1 second320 bits * 50 = 16,000 bits/sec, or 16 kbps
Voice packets are benign compared to the bandwidth consumed by data applications. Data applications can fill the entire maximum transmission unit (MTU) of an Ethernet frame (1518 bytes or 9216 bytes if jumbo Ethernet frames maintain been enabled). In comparison to data application packets, voice packets are very little (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling rate).
In pattern 1-2, a conference bridge has been deployed at the main site. No conference bridge exists at the remote site. If three IP Phones at a remote site unite a conference, their RTP streams are sent across the WAN to the conference bridge. The conference bridge, whether using software or hardware resources, mixes the received audio streams and then sends back three unique unicast audio streams to the IP Phones over the IP WAN. The conference bridge removes the receiver's voice from his or her unique RTP stream so that the user does not suffer echo because of the delay of traversing the WAN link and mixing RTP audio streams in the conference bridge.
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Centralized conference resources cause bandwidth, delay, and capacity challenges in the voice network. Each G.711 RTP stream requires 80 kbps (plus the Layer 2 overhead), resulting in 240 kbps of IP WAN bandwidth consumption by this voice conference. If the conference bridge were not located on the other side of the IP WAN, this traffic would not exigency to traverse the WAN link, resulting in less delay and bandwidth consumption. If the remote site had a CUCM region configuration that resulted in calls with the G.729 codec back to the main site, the software conferencing resources of CUCM would not subsist able to coalesce the audio conversations. Hardware conferencing or hardware transcoder media resources in a voice gateway are required to accommodate G.729 audio conferencing. Local hardware conference resources would remove this need. every sole centrally located media resources (Music On Hold [MOH], annunciator, conference bridges, videoconferencing, and media termination points) suffer similar bandwidth, delay, and resource exhaustion challenges.
When deploying CUCM in multisite environments, centralized CUCM-based services are accessed over the IP WAN. Affected services include the following:
Figure 1-3 shows a Unified Communications network in which the main site is connected to a remote site through a centralized call-processing environment. The main site is besides connected to a remote cluster through an intercluster trunk (ICT) representing a distributed call processing environment. The combination of both centralized and distributed call processing represents a hybrid call-processing model in which little sites exercise the CUCM resources of the main site, but large remote offices maintain their own CUCM cluster. On the bottom left of pattern 1-3 is a SIP trunk, typically over a Metro Ethernet connection to an Internet Telephony Service Provider (ITSP). The profit of the SIP trunk is that the ITSP provides the gateways to the PSTN instead of your providing gateways at the main site.
Click for a larger image
An IP WAN outage in pattern 1-3 will cause an outage of call-processing services for the remote site connected in a centralized fashion. The remote cluster will not suffer a call-processing outage, but the remote cluster will not subsist able to dial the main site over the IP WAN during the outage. Mission-critical voice applications (voice mail, interactive voice response [IVR], and so on) located at the main site will subsist unavailable to any of the other sites during the WAN outage.
If the ITSP is using the selfsame links that allow IP WAN connectivity, every sole calls to and from the public switched telephone network (PSTN) will besides subsist unavailable.
A deployment enjoy the one shown in pattern 1-3 is considered badly designed because of the exigency of IP WAN and PSTN backup.
Dial scheme challenges
In a multisite deployment, with a sole or multiple CUCM clusters, dial scheme design requires the consideration of several issues that conclude not exist in single-site deployments:
Each internal directory number relates to a fixed-length PSTN number: In this case, each internal directory number has its own dedicated PSTN number. The directory number can, but does not maintain to, match the least-significant digits of the PSTN number. In countries with a fixed numbering plan, such as the North American Numbering scheme (NANP), this usually means that the four-digit office codes are used as internal directory numbers. If these are not unique, digits of office codes or administratively assigned site codes might subsist added, resulting in five or more digits being used for internal directory numbers.
Another solution is to not reuse any digits of the PSTN number but to simply map each internally used directory number to any PSTN number assigned to the company. In this case, the internal and external numbers conclude not maintain anything in common. If the internally used directory number matches the least-significant digits of its corresponding PSTN number, significant digits can subsist set at the gateway or trunk. Also, generic external phone number masks, transformation masks, or prefixes can subsist configured. This is just because every sole internal directory numbers are changed to fully qualified PSTN numbers in the selfsame way. Another example is if the internal directory number is composed of parts of the PSTN number and administratively assigned digits such as site codes plus PSTN station codes, or different ranges, such as PSTN station codes 4100 to 4180 that map to directory numbers 1100 to 1180, or totally independent mappings of internal directory numbers to PSTN numbers. In that case, one or more translation rules maintain to subsist used for incoming calls, and one or more calling party transformation rules, transformation masks, external phone number masks, or prefixes maintain to subsist configured.
Type of Number (TON) in ISDN: The calling number (the Automatic Number Identification [ANI]) of calls being received from the PSTN can subsist represented in different ways:
- As a seven-digit subscriber number- As a ten-digit number, including the area code- In international format with the country code in front of the area code
To standardize the ANI for every sole calls, the format that is used must subsist known, and the number has to subsist transformed accordingly.
Overlapping and nonconsecutive numbers
In pattern 1-4, Cisco IP Phones at the main site exercise directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. At the remote site, 1001 to 1099 and 2158 to 2364 are used. These directory numbers maintain two issues. First, 1001 to 1099 overlap; these directory numbers exist at both sites, so they are not unique throughout the complete deployment. This causes a problem: If a user in the remote site dialed only the four digits 1001, which phone would ring? This issue of overlapping dial plans needs to subsist addressed by digit manipulation. In addition, the nonconsecutive exercise of the purview 2000 to 2999 (with some duplicate numbers at the two sites) would require a significant number of additional entries in call-routing tables because the ranges can hardly subsist summarized by one (or a few) entries.
Click for a larger image
Fixed versus variable-length numbering plans
A fixed numbering scheme features fixed-length area codes and local numbers. An open numbering scheme features variance in length of area code or local number, or both, within the country.
Table 1-1 contrasts the NANP and a variable-length numbering scheme -- Germany's numbering scheme in this example.
Table 1-1Component Description Fixed Numbering scheme (NANP) Variable-Length Numbering scheme (Germany) Country code A code of one to three digits is used to achieve the particular telephone system for each nation or special service. Obtain the E.164 benchmark from http://itu.org to observe every sole international country codes. 1 49 Area code Used within many nations to route calls to a particular city, region, or special service. Depending on the nation or region, it may besides subsist called a numbering scheme area, subscriber trunk dialing code, national destination code, or routing code. Three digits Three to five digits Subscriber number Represents the specific telephone number to subsist dialed, but it does not include the country code, area code (if applicable), international prefix, or trunk prefix. Three-digit exchange code plus a four-digit station code Three or more digits Trunk prefix The initial digits to subsist dialed in a domestic call, before the area code and the subscriber number. 1 0 Access code A number that is traditionally dialed first "to acquire out to the PSTN," used in PBXs and VoIP systems. 9 0 International prefix The code dialed before an international number (country code, area code if any, and then subscriber number). 011 00 or + (+ is used by cell phones)
The NANP PSTN number is 408-555-1234, DID is not used, and every sole calls placed to the main site are handled by an attendant. There is a remote site in Germany with the E.164 PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can subsist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. Note how the + is replaced by the international prefix 011 and the access code 9. If the phone with extension 1001 should subsist called directly, 9-011-49-404-13267-1001 has to subsist dialed.
Variable-length numbering, E.164 addressing and DID
Figure 1-5 illustrates an example in which the main site with CUCM resides in the U.S. and a remote site without CUCM resides in Germany. The NANP PSTN number in the U.S. is 408-555-1234. Note that DID is not used, because every sole calls placed to the main site are handled by an attendant. A remote site in Germany has PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can subsist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. If the phone with extension 1001 should subsist called directly, 9-011-49-404-13267-1001 has to subsist dialed.
Click for a larger image
The logic of routing calls by CUCM over the WAN or through the PSTN is appropriately transparent to the phone user.
Optimized call routing and PSTN backup
There are two ways to save costs for PSTN calls in a multisite deployment:
In the example shown in pattern 1-6, a call from Chicago to San Jose would subsist routed as follows:
Click for a larger image
If the WAN were unavailable for any reason before the call, the Chicago Gateway would maintain to subsist properly configured to route the call with the appropriate digit manipulation through the PSTN at a potentially higher toll cost to the San Jose PSTN phone.
NAT and security issues
In single-site deployments, CUCM servers and IP Phones usually exercise private IP addresses because there is no exigency to communicate with the outside IP world. NAT is not configured for the phone subnets, and attacks from the outside are impossible. In multisite deployments, however, IP Security (IPsec) virtual private network (VPN) tunnels can subsist used between sites. The VPN tunnels allow only intersite communication; access to the protected internal networks is not practicable from the outside—only from the other site through the tunnel. Therefore, attacks from the outside are blocked at the gateway. To configure IPsec VPNs, the VPN tunnel must subsist configured to terminate on the two gateways in the different sites. Sometimes this is not possible; for instance, the two sites may subsist under different administration, or perhaps security policies conclude not allow the configuration of IPsec VPNs.
In such a case, or when connecting to a public service such as an ITSP, NAT has to subsist configured for CUCM servers and IP Phones. Cisco calls this Hosted NAT Traversal for Session edge Controllers.
In pattern 1-7, Company A and Company B both exercise IP network 10.0.0.0/8 internally. To communicate over the Internet, the private addresses are translated into public IP addresses. Company A uses public IP network A, and Company B uses public IP network B. every sole CUCM servers and IP Phones can subsist reached from the Internet and communicate with each other.
As soon as CUCM servers and IP Phones can subsist reached with public IP addresses, they are topic to attacks from the outside world, introducing potential security issues.
Click for a larger imageThis download first appeared at searchunifiedcommunications.com
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